IP Multimedia services provide a dynamic combination of voice, video, messaging, data, etc, within the same session. This has lead to a growth in the numbers of basic applications and the media which it is possible to combine, leading to a growth in the number and variety of services offered to the end users—so-called “combinational IP Multimedia” services.
IP Multimedia Subsystem (IMS) is the technology defined by the Third Generation Partnership Project (3GPP) to provide IP Multimedia services over mobile communication networks. IMS provides key features to enrich the end-user person-to-person communication experience through the integration and interaction of services. IMS allows new rich person-to-person (client-to-client) as well as person-to-content (client-to-server) communications over an IP-based network. The IMS makes use of the Session Initiation Protocol (SIP) to set up and control calls or sessions between user terminals (or user terminals and application servers). Whilst SIP was created as a user-to-user protocol, IMS allows operators and service providers to control user access to services and to charge users accordingly. Other protocols are used for media transmission and control, such as Real-time Transport Protocol and Real-time Transport Control Protocol (RTP/RTCP).
FIG. 1 illustrates schematically how the IMS fits into the mobile network architecture in the case of a General Packet Radio Service (GPRS) access network. As shown in FIG. 1 control of communications occurs at three layers (or planes). The lowest layer is the Connectivity Layer 1, also referred to as the bearer plane and through which signals are directed to/from user equipment (UE) accessing the network. The entities within the connectivity layer 1 that connect an IMS subscriber to IMS services form a network that is referred to as the IP-Connectivity Access Network, IP-CAN. The GPRS network includes various GPRS Support Nodes (GSNs). A gateway GPRS support node (GGSN) 2 acts as an interface between the GPRS backbone network and other networks (radio network and the IMS network). The middle layer is the Control Layer 4, and at the top is the Application Layer 6.
The IMS 3 includes a core network 3a, which operates over the middle, Control Layer 4 and the Connectivity Layer 1, and a Service Network 3b. The IMS core network 3a includes nodes that send/receive signals to/from the GPRS network via the GGSN 2 at the Connectivity Layer 1 and network nodes that include Call/Session Control Functions (CSCFs) 5, which operate as SIP proxies within the IMS in the middle, Control Layer 4. The top, Application Layer 6 includes the IMS service network 3b. Application Servers (ASs) 7 are provided for implementing IMS service functionality.
As shown in FIG. 1, User Equipment (UE) can access the IMS by attaching to an access network and then over the Connectivity Layer 1, which is part of a PS domain. In that case an IMS session can be set up by the UE using SIP signalling. FIG. 2 illustrates schematically the main components that are relevant to the present disclosure of a PS Evolved Packet Core (EPC) network in accordance with the 3GPP defined Systems Architecture Evolution (SAE), and shows a UE 20 accessing an IP network shown as the Internet 21. The principal network entities shown include a Serving Gateway (SGW) 23, a PDN Gateway (PGW) 24, an evolved NodeB (eNodeB) 25, a Mobility Management Entity (MME) 26 and the user's Home Subscriber Server (HSS) 27. For the purposes of the following discussion the SGW 23 and PGW 24 will be grouped together as one entity SGW/PGW.
Many existing access networks operate only using CS technology, but a UE may also access IMS services via a CS domain. Although the CS domain will not handle SIP, procedures are well established for dealing with the provision of media and services between the IMS and a UE using a CS access. There are many occasions when during a call/session it is required to transfer or hand over the call/session from one access network to another. There are a variety of factors that are used to determine when a call needs to be handed over to another access network, but these are not particularly relevant to the present discussion. All we need to know is that the CS access network determines, based on the cells for which the UE reports measurements, when the conditions arise that require a request to be made to the core network for the call to be handed over. Single Radio Voice Call Continuity (SRVCC) is described in 3GPP TS 23.237 and 3GPP TS 23.216, which specify procedures for handover of a voice call from a PS access to a CS access (e.g. transfer of a VoIP IMS session from an E-UTRAN to a UTRAN/GERAN). These technical specifications have also been extended to allow handover of a voice call from a CS access to a PS access.
However, the procedures specified for CS to PS handover with SRVCC do not specify any procedures for the transfer of an emergency call. As the emergency call in the CS domain has already been established to a particular destination Public Safety Access Point (PSAP), it is important to ensure that the emergency call towards the PSAP is not interrupted when the transfer to the PS domain occurs. This means that the PSAP must not be changed or impacted in a way that could interrupt the call. Once the routing of the emergency call to the PSAP has been set up, the handover procedures need to ensure that the mobility is hidden from the PSAP. Also, there may be special considerations for handling emergency calls that ensure the call is properly routed and not delayed or interrupted. Under the current procedures, the special considerations that are put in place for an emergency call in the CS domain may not be translated into corresponding considerations in the PS domain when the call is transferred. The transferred call could lose its emergency call status. This problem is addressed by the following discussion.